Many two-way radio products today operate using both analog and digital modulation for voice modes. For example, the Association of Public Safety Communications Officials (APCO) 25 radio standard utilizes both standard analog frequency modulation (FM) and frequency division multiple access (FDMA) digital modulation. In practice, when the radio transceiver is switching between analog and digital modes, users listening to this radio may perceive changes in microphone input level. This manifests itself in the form of audio output signal levels having both high and low amplitudes. In the past, in order to prevent the listener from continually changing volume levels to compensate for this variance, the transmitter microphone input level was balanced by setting fixed gain levels in both the transmit and receive audio paths. This approach however has not always been effective, leading to an inconsistent or non-uniform audio output.
As seen in FIG. 1, when the received volume or audio output speaker level is plotted versus the transmitted or microphone input speaker level in an analog mode 101, the amplitude response curve is very non-linear. This non-linear shape results from the fact that audio is typically compressed while operating in an analog mode resulting in non-linear microphone audio gain. While in an analog mode, the system deviation can typically be set at approximately 60 percent of the maximum at an input level of approximately 95 decibel (dB) speaker pressure level (SPL). Therefore when the audio input signal level is greater than this level, the system deviation is limited by clipping at a preset level. This has the effect of compressing the amplitude of the transmitted analog audio signal leading to an analog volume curve 101 as seen in FIG. 1, which creates a non-linear response. In practice, this results in a system dynamic range of only a few dB while in an analog mode.
In a digital mode 103, there is no clipping circuit to limit maximum deviation, since the transmitted audio information is digitally encoded. Thus, digital mode transmissions have a much higher dynamic range than analog transmissions. Moreover, voice encoders or “vocoders” used in the digital mode encode digital audio and do not tolerate a compressed signal well. The vocoder tends to degrade audio quality when beyond a predetermined input level. These facts lead the digital transmit audio being linear instead of compressed as in the analog mode.
Consequently, these variations between audio in the analog and digital modes typically result in field complaints in audio output level in radio products. Users perceive that a radio is not operating properly since the volume levels in the analog and digital modes must be continually adjusted in order to achieve a constant amplitude level. Users may also complain that the digital mode is not tolerant of microphone input variations in mouth-to-speaker distances as it is while in the analog since compression tends to be compensate for variation in input levels.
In other words, the audio level in the digital modes is reduced at a greater rate as the user moves further from the microphone. This ultimately reduces microphone sensitivity below a users desired specifications. Further issues are created related to unintelligible audio at high volume levels when in the digital mode. This is due to the large dynamic range entering in to a “clip” or distortion where the analog mode is more forgiving and acts as a pseudo-automatic gain control by limiting the audio input level. Using a fixed gain to adjust one signal will only match the modes at one point.
Accordingly, the need exists to provide a method for the efficient control for audio microphone gain balance in two-way communications equipment operating in both an analog and digital modulation mode.